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3rd Edition: Chapter 3

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Title: 3rd Edition: Chapter 3


1
Transport Layer
Adapted from Computer Networking slides
2
Transport services and protocols
  • provide logical communication between app
    processes
  • End system TCP or UDP
  • sending side breaks app messages into segments,
    passes to network layer
  • receiving side reassembles segments into
    messages, passes to app layer

3
Transport vs. network layer
  • network layer logical communication between
    hosts
  • transport layer logical communication between
    processes
  • relies on and enhances - network layer services

4
Internet transport-layer protocols
  • reliable, in-order delivery (TCP)
  • congestion control
  • flow control
  • connection setup
  • unreliable, unordered delivery UDP
  • no-frills extension of best-effort IP
  • services not available
  • delay guarantees
  • bandwidth guarantees

5
Multiplexing/demultiplexing
delivering received segments to correct socket
gathering data from multiple sockets, enveloping
data with header
process
socket
application
P4
application
application
P1
P2
P3
P1
transport
transport
transport
network
network
network
link
link
link
physical
physical
physical
host 3
host 2
host 1
6
How demultiplexing works
  • host receives IP datagrams
  • each datagram has source IP address, destination
    IP address
  • each datagram carries one transport-layer segment
  • each segment has source port and destination port
    number
  • host uses IP addresses port numbers to direct
    segment to appropriate socket

32 bits
source port
dest port
other header fields
application data (message)
TCP/UDP segment format
7
Connectionless demultiplexing
  • When host receives UDP segment
  • checks destination port number in segment
  • directs UDP segment to socket with that port
    number
  • IP datagrams with different source IP addresses
    and/or source port numbers directed to same
    socket
  • Create sockets with port numbers
  • DatagramSocket mySocket1 new DatagramSocket(8000
    )
  • DatagramSocket mySocket2 new DatagramSocket(9000
    )
  • UDP socket identified by two-tuple
  • (dest IP address, dest port number)

8
Connectionless demux (cont)
  • DatagramSocket serverSocket new
    DatagramSocket(6428)

Source Port provides return address
9
Connection-oriented demux
  • TCP socket identified by 4-tuple
  • source IP address
  • source port number
  • dest IP address
  • dest port number
  • recv host uses all four values to direct segment
    to appropriate socket
  • Server host may support many simultaneous TCP
    sockets
  • each socket identified by its own 4-tuple
  • Web servers have different sockets for each
    connecting client
  • non-persistent HTTP will have different socket
    for each request

10
Connection-oriented demux (cont)
S-IP B
D-IPC
SP 9157
Client IPB
DP 80
server IP C
S-IP A
S-IP B
D-IPC
D-IPC
11
Connection-oriented demux Threaded Web Server
P4
S-IP B
D-IPC
SP 9157
Client IPB
DP 80
server IP C
S-IP A
S-IP B
D-IPC
D-IPC
12
UDP User Datagram Protocol RFC 768
  • no frills, bare bones Internet transport
    protocol
  • best effort service, UDP segments may be
  • lost
  • delivered out of order to app
  • connectionless
  • no handshaking between UDP sender, receiver
  • each UDP segment handled independently of others
  • Why is there a UDP?
  • no connection establishment (which can add delay)
  • simple no connection state at sender, receiver
  • small segment header
  • no congestion control UDP can blast away as fast
    as desired

13
UDP more
  • often used for streaming multimedia apps
  • loss tolerant
  • rate sensitive
  • other UDP uses
  • DNS
  • SNMP
  • reliable transfer over UDP add reliability at
    application layer
  • application-specific error recovery!

32 bits
source port
dest port
Length, in bytes of UDP segment, including header
checksum
length
Application data (message)
UDP segment format
14
UDP checksum
  • Goal detect errors (e.g., flipped bits) in
    transmitted segment
  • Sender
  • treat segment contents as sequence of 16-bit
    integers
  • checksum addition (1s complement of sum) of
    segment contents
  • sender puts checksum value into UDP checksum
    field
  • Receiver
  • compute checksum of received segment
  • check if computed checksum equals checksum field
    value
  • NO - error detected
  • YES - no error detected. But maybe errors
    nonetheless? More later .

15
Internet Checksum Example
  • Note
  • When adding numbers, a carryout from the most
    significant bit needs to be added to the result
  • Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1
0 1 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0
1 1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1
1 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1
0 0 1 0 1 0 0 0 1 0 0 0 1 0 0 0 0
1 1
wraparound
sum
checksum
16
Principles of Reliable data transfer
  • important in app., transport, link layers
  • top-10 list of important networking topics!
  • characteristics of unreliable channel will
    determine complexity of reliable data transfer
    protocol (rdt)

17
Principles of Reliable data transfer
  • important in app., transport, link layers
  • top-10 list of important networking topics!
  • characteristics of unreliable channel will
    determine complexity of reliable data transfer
    protocol (rdt)

18
Principles of Reliable data transfer
  • important in app., transport, link layers
  • top-10 list of important networking topics!
  • characteristics of unreliable channel will
    determine complexity of reliable data transfer
    protocol (rdt)

19
Reliable data transfer getting started
send side
receive side
20
Rdt1.0 reliable transfer over a reliable channel
  • underlying channel perfectly reliable
  • no bit errors
  • no loss of packets
  • separate FSMs for sender, receiver
  • sender sends data into underlying channel
  • receiver read data from underlying channel

rdt_send(data)
rdt_rcv(packet)
Wait for call from below
Wait for call from above
extract (packet,data) deliver_data(data)
packet make_pkt(data) udt_send(packet)
sender
receiver
21
Rdt2.0 channel with bit errors
  • underlying channel may flip bits in packet
  • checksum to detect bit errors
  • the question how to recover from errors
  • acknowledgements (ACKs) receiver explicitly
    tells sender that pkt received OK
  • negative acknowledgements (NAKs) receiver
    explicitly tells sender that pkt had errors
  • sender retransmits pkt on receipt of NAK

22
rdt2.0 FSM specification
rdt_send(data)
receiver
snkpkt make_pkt(data, checksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) isNAK(rcvpkt)
Wait for call from above
udt_send(sndpkt)
rdt_rcv(rcvpkt) isACK(rcvpkt)
L
sender
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(A
CK)
23
rdt2.0 FSM specification
rdt_send(data)
receiver
snkpkt make_pkt(data, checksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) isNAK(rcvpkt)
Wait for call from above
udt_send(sndpkt)
rdt_rcv(rcvpkt) isACK(rcvpkt)
L
sender
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
  • Problem!
  • What happens if ACK/NAK corrupted?

extract(rcvpkt,data) deliver_data(data) udt_send(A
CK)
24
rdt2.1 sender, handles garbled ACK/NAKs
rdt_send(data)
sndpkt make_pkt(0, data, checksum) udt_send(sndp
kt)
rdt_rcv(rcvpkt) ( corrupt(rcvpkt)
isNAK(rcvpkt) )
Wait for call 0 from above
udt_send(sndpkt)
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
isACK(rcvpkt)
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
isACK(rcvpkt)
L
L
rdt_rcv(rcvpkt) ( corrupt(rcvpkt)
isNAK(rcvpkt) )
rdt_send(data)
sndpkt make_pkt(1, data, checksum) udt_send(sndp
kt)
udt_send(sndpkt)
25
rdt2.1 receiver, handles garbled ACK/NAKs
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
has_seq0(rcvpkt)
extract(rcvpkt,data) deliver_data(data) sndpkt
make_pkt(ACK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) (corrupt(rcvpkt)
rdt_rcv(rcvpkt) (corrupt(rcvpkt)
sndpkt make_pkt(NAK, chksum) udt_send(sndpkt)
sndpkt make_pkt(NAK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) not corrupt(rcvpkt)
has_seq1(rcvpkt)
rdt_rcv(rcvpkt) not corrupt(rcvpkt)
has_seq0(rcvpkt)
sndpkt make_pkt(ACK, chksum) udt_send(sndpkt)
sndpkt make_pkt(ACK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
has_seq1(rcvpkt)
extract(rcvpkt,data) deliver_data(data) sndpkt
make_pkt(ACK, chksum) udt_send(sndpkt)
26
rdt2.2 sender, receiver fragments
rdt_send(data)
sndpkt make_pkt(0, data, checksum) udt_send(sndp
kt)
rdt_rcv(rcvpkt) ( corrupt(rcvpkt)
isACK(rcvpkt,1) )
udt_send(sndpkt)
sender FSM fragment
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
isACK(rcvpkt,0)
rdt_rcv(rcvpkt) (corrupt(rcvpkt)
has_seq1(rcvpkt))
L
receiver FSM fragment
udt_send(sndpkt)
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
has_seq1(rcvpkt)
extract(rcvpkt,data) deliver_data(data) sndpkt
make_pkt(ACK1, chksum) udt_send(sndpkt)
27
rdt3.0 channels with errors and loss
  • New assumption underlying channel can also lose
    packets (data or ACKs)
  • Approach sender waits reasonable amount of
    time for ACK
  • retransmits if no ACK received in this time
  • requires countdown timer
  • if pkt (or ACK) just delayed (not lost) What
    then?

sender
recevier
28
rdt3.0 sender
rdt_send(data)
rdt_rcv(rcvpkt) ( corrupt(rcvpkt)
isACK(rcvpkt,1) )
sndpkt make_pkt(0, data, checksum) udt_send(sndp
kt) start_timer
L
rdt_rcv(rcvpkt)
L
timeout
udt_send(sndpkt) start_timer
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
isACK(rcvpkt,1)
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
isACK(rcvpkt,0)
stop_timer
stop_timer
timeout
udt_send(sndpkt) start_timer
rdt_rcv(rcvpkt)
L
rdt_send(data)
rdt_rcv(rcvpkt) ( corrupt(rcvpkt)
isACK(rcvpkt,0) )
sndpkt make_pkt(1, data, checksum) udt_send(sndp
kt) start_timer
L
29
rdt3.0 in action
30
rdt3.0 in action
31
Performance of rdt3.0
  • rdt3.0 works, but performance stinks
  • example 1 Gbps link, 15 ms e-e prop. delay, 1KB
    packet

L (packet length in bits)
8kb/pkt
T


.008 msec
transmit
R (transmission rate, bps)
109 b/sec
  • U sender utilization fraction of time sender
    busy sending
  • 1KB pkt every 30 msec -gt 33kB/sec thruput over 1
    Gbps link
  • network protocol limits use of physical resources!

32
rdt3.0 stop-and-wait operation
sender
receiver
first packet bit transmitted, t 0
last packet bit transmitted, t L / R
first packet bit arrives
RTT
last packet bit arrives, send ACK
ACK arrives, send next packet, t RTT L / R
33
Pipelined protocols
  • Pipelining sender allows multiple, in-flight,
    yet-to-be-acknowledged pkts
  • range of sequence numbers must be increased
  • buffering at sender and/or receiver
  • Two generic forms of pipelined protocols
    go-Back-N, selective repeat

34
Pipelining increased utilization
sender
receiver
first packet bit transmitted, t 0
last bit transmitted, t L / R
first packet bit arrives
RTT
last packet bit arrives, send ACK
last bit of 2nd packet arrives, send ACK
last bit of 3rd packet arrives, send ACK
ACK arrives, send next packet, t RTT L / R
Increase utilization by a factor of 3!
35
Go-Back-N
  • Sender
  • k-bit seq in pkt header
  • window of up to N, consecutive unacked pkts
    allowed
  • Cumulative ACK ( Ack(n) ) ACKs all pkts up to,
    including seq n
  • may receive duplicate ACKs (see receiver)
  • timer for each in-flight pkt
  • timeout(n) retransmit pkt n and all higher seq
    pkts in window

36
GBN sender extended FSM
rdt_send(data)
if (nextseqnum lt baseN) sndpktnextseqnum
make_pkt(nextseqnum,data,chksum)
udt_send(sndpktnextseqnum) if (base
nextseqnum) start_timer nextseqnum
else refuse_data(data)
L
base1 nextseqnum1
timeout
start_timer udt_send(sndpktbase) udt_send(sndpkt
base1) udt_send(sndpktnextseqnum-1)
rdt_rcv(rcvpkt) corrupt(rcvpkt)
L
rdt_rcv(rcvpkt) notcorrupt(rcvpkt)
base getacknum(rcvpkt)1 If (base
nextseqnum) stop_timer else start_timer
37
GBN receiver extended FSM
default
udt_send(sndpkt)
rdt_rcv(rcvpkt) notcurrupt(rcvpkt)
hasseqnum(rcvpkt,expectedseqnum)
L
Wait
extract(rcvpkt,data) deliver_data(data) sndpkt
make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpk
t) expectedseqnum
expectedseqnum1 sndpkt
make_pkt(expectedseqnum,ACK,chksum)
  • ACK-only always send ACK for correctly-received
    pkt with highest in-order seq
  • may generate duplicate ACKs
  • need only remember expectedseqnum
  • out-of-order pkt
  • discard (dont buffer) -gt no receiver buffering!
  • Re-ACK pkt with highest in-order seq

38
GBN in action
39
Selective Repeat
Less traffic - because less resend activity
  • receiver individually acknowledges all correctly
    received pkts
  • buffers pkts, as needed, for eventual in-order
    delivery to upper layer
  • sender only resends pkts for which ACK not
    received
  • sender timer for each unACKed pkt
  • sender window
  • N consecutive seq s
  • again limits seq s of sent, unACKed pkts

40
Selective repeat sender, receiver windows
41
Selective repeat
  • pkt n in rcvbase, rcvbaseN-1
  • send ACK(n)
  • out-of-order buffer
  • in-order deliver (also deliver buffered,
    in-order pkts), advance window to next
    not-yet-received pkt
  • pkt n in rcvbase-N,rcvbase-1
  • ACK(n)
  • correctly received - previously acknowledged
  • otherwise
  • ignore
  • data from above
  • if next available seq in window, send pkt
  • timeout(n)
  • resend pkt n, restart timer
  • ACK(n) in sendbase,sendbaseN
  • mark pkt n as received
  • if n smallest unACKed pkt, advance window base to
    next unACKed seq

42
Selective repeat in action
43
Selective repeat dilemma
  • Example
  • seq s 0, 1, 2, 3
  • window size3
  • receiver sees no difference in two scenarios!
  • incorrectly passes duplicate data as new in (a)
  • Q what relationship between seq size and
    window size?

44
TCP Overview RFCs 793, 1122, 1323, 2018, 2581
  • point-to-point
  • one sender, one receiver
  • reliable, in-order byte steam
  • no message boundaries
  • pipelined
  • TCP congestion and flow control set window size
  • send receive buffers
  • full duplex data
  • bi-directional data flow in same connection
  • MSS maximum segment size
  • connection-oriented
  • handshaking (exchange of control msgs) inits
    sender, receiver state before data exchange
  • flow controlled
  • sender will not overwhelm receiver

45
TCP segment structure
URG urgent data (generally not used)
counting by bytes of data (not segments!)
ACK ACK valid
PSH push data now (generally not used)
bytes rcvr willing to accept
RST, SYN, FIN connection estab (setup,
teardown commands)
Internet checksum (as in UDP)
46
TCP seq. s and ACKs
  • Seq. s
  • byte stream number of first byte in segments
    data
  • ACKs
  • seq of next byte expected from other side
  • cumulative ACK
  • Q how receiver handles out-of-order segments
  • A TCP spec doesnt say, - up to implementor

Host B
Host A
User types C
Seq42, ACK79, data C
host ACKs receipt of C, echoes back C
Seq79, ACK43, data C
host ACKs receipt of echoed C
Seq43, ACK80
simple telnet scenario
47
TCP Round Trip Time and Timeout
  • Q how to estimate RTT?
  • SampleRTT measured time from segment
    transmission until ACK receipt
  • ignore retransmissions
  • SampleRTT will vary, want estimated RTT
    smoother
  • average several recent measurements, not just
    current SampleRTT
  • Q how to set TCP timeout value?
  • longer than RTT (varies)
  • too short premature timeout
  • Unnecessary retransmissions
  • too long slow reaction to segment loss

48
TCP Round Trip Time and Timeout
EstimatedRTT (1- ?)EstimatedRTT ?SampleRTT
  • Exponential weighted moving average
  • influence of past sample decreases exponentially
    fast
  • typical value ? 0.125

Sample EstimatedRTT
49
TCP Round Trip Time and Timeout
  • Setting the timeout
  • EstimtedRTT plus safety margin
  • large variation in EstimatedRTT -gt larger safety
    margin
  • first estimate of how much SampleRTT deviates
    from EstimatedRTT

DevRTT (1-?)DevRTT
?SampleRTT-EstimatedRTT (typically, ? 0.25)
Safety margin
Then set timeout interval
TimeoutInterval EstimatedRTT 4DevRTT
50
TCP reliable data transfer
REAL
  • Created on top of IPs unreliable service
  • Pipelined segments
  • Cumulative acks
  • TCP uses single retransmission timer
  • Retransmissions are triggered by
  • timeout events
  • duplicate acks

51
TCP sender events
  • data rcvd from app
  • Create segment with seq
  • seq is byte-stream number of first data byte in
    segment
  • start timer if not already running (think of
    timer as for oldest unacked segment)
  • expiration interval TimeOutInterval
  • timeout
  • retransmit segment that caused timeout
  • restart timer
  • Ack rcvd
  • If acknowledges previously unacked segments
  • update what is known to be acked
  • start timer if there are outstanding segments

52
TCP sender (simplified)
NextSeqNum InitialSeqNum
SendBase InitialSeqNum loop (forever)
switch(event) event
data received from application above
create TCP segment with sequence number
NextSeqNum if (timer currently
not running) start timer
pass segment to IP
NextSeqNum NextSeqNum length(data)
event timer timeout
retransmit not-yet-acknowledged segment with
smallest sequence number
start timer event ACK
received, with ACK field value of y
if (y gt SendBase)
SendBase y if (there are
currently not-yet-acknowledged segments)
start timer
/ end of loop forever /
Example SendBase-1 71 last acked byte y 73,
so the rcvr wants 73 y gt SendBase, so that new
data is acked
Ignore duplicate ACKs
53
TCP retransmission scenarios
Host A
Host B
Seq92, 8 bytes data
Seq100, 20 bytes data
ACK100
ACK120
Seq92, 8 bytes data
Sendbase 100
SendBase 120
ACK120
Seq92 timeout
SendBase 100
SendBase 120
premature timeout
54
TCP retransmission scenarios
SendBase 120
55
TCP ACK generation RFC 1122, RFC 2581
TCP Receiver action Delayed ACK. Wait up to
500ms for next segment. If no next segment, send
ACK Immediately send single cumulative ACK,
ACKing both in-order segments Immediately send
duplicate ACK, indicating seq. of next
expected byte Immediately send ACK, provided
that segment starts at lower end of gap
Event at Receiver Arrival of in-order segment
with expected seq . All data up to expected seq
already ACKed Arrival of in-order segment
with expected seq . One other segment has ACK
pending Arrival of out-of-order
segment higher-than-expect seq. . Gap
detected Arrival of segment that partially or
completely fills gap
56
Fast Retransmit
  • Time-out period often relatively long
  • long delay before resending lost packet
  • Detect lost segments via duplicate ACKs.
  • Sender pattern many segments back-to-back
  • If segment is lost - many duplicate ACKs.
  • If sender receives 3 ACKs for the same data, it
    supposes that segment after ACKed data was lost
  • fast retransmit resend segment before timer
    expires

57
Fast retransmit algorithm
Switch out last event in sender algorithm
event ACK received, with ACK field value of y
if (y gt SendBase)
SendBase y
if (there are currently not-yet-acknowledged
segments) start
timer
else increment count
of dup ACKs received for y
if (count of dup ACKs received for y 3)
resend segment with
sequence number y

a duplicate ACK for already ACKed segment
fast retransmit
58
TCP Flow Control
  • receive side of TCP connection has a receive
    buffer
  • speed-matching service matching the send rate to
    the receiving apps drain rate
  • app process may be slow at reading from buffer

59
TCP Flow control how it works
  • Rcvr advertises spare room by including value of
    RcvWindow in segments
  • Sender limits unACKed data to RcvWindow
  • guarantees receive buffer doesnt overflow

Circular buffer
  • (Suppose TCP receiver discards out-of-order
    segments)
  • spare room in buffer
  • RcvWindow RcvBuffer -LastByteRcvd
    -LastByteRead

No Flow control in UDP
60
TCP Connection Management
  • Three way handshake
  • Step 1 client host sends TCP SYN segment to
    server
  • specifies initial seq
  • no data
  • Step 2 server host receives SYN, replies with
    SYNACK segment
  • server allocates buffers
  • specifies server initial seq.
  • Step 3 client receives SYNACK, replies with ACK
    segment, which may contain data
  • Recall TCP sender, receiver establish
    connection before exchanging data segments
  • initialize TCP variables
  • seq. s
  • buffers, flow control info (e.g. RcvWindow)
  • client connection initiator
  • Socket clientSocket new Socket("hostname","p
    ort number")
  • server contacted by client
  • Socket connectionSocket welcomeSocket.accept()

61
TCP Connection Management
  • Closing a connection
  • client closes socket clientSocket.close()
  • Step 1 client end system sends TCP FIN control
    segment to server
  • Step 2 server receives FIN, replies with ACK.
    Closes connection, sends FIN.

62
TCP Connection Management
  • Step 3 client receives FIN, replies with ACK.
  • Enters timed wait - will respond with ACK to
    received FINs
  • Step 4 server, receives ACK. Connection closed.
  • Note with small modification, can handle
    simultaneous FINs.

client
server
closing
FIN
ACK
closing
FIN
ACK
timed wait
closed
closed
63
TCP Connection Management
TCP server lifecycle
TCP client lifecycle
64
Principles of Congestion Control
  • Congestion
  • informally too many sources sending too much
    data too fast for a network to handle
  • different from flow control!
  • manifestations
  • lost packets (buffer overflow at routers)
  • long delays (queueing in router buffers)

65
Causes/costs of congestion scenario 1
  • two senders, two receivers
  • one router, infinite buffers
  • no retransmission

Ideal steady work for link
  • large delays when congested
  • maximum achievable throughput

R/2
R/2
R/2
Bytes/sec
66
Causes/costs of congestion scenario 2
  • one router, finite buffers
  • sender retransmission of lost packet

Host A
lout
lin original data
l'in original data, plus retransmitted data
Host B
finite shared output link buffers
67
Causes/costs of congestion scenario 2
  • always (goodput)
  • perfect retransmission only when loss
  • retransmission of delayed (not lost) timeout
    prematurally retransmit unnecessary

realistic
realistic
  • costs of congestion
  • more work (retrans) for given goodput
  • unneeded retransmissions link carries multiple
    copies of pkt

68
Causes/costs of congestion scenario 3
  • four senders
  • multihop paths
  • timeout/retransmit

Q what happens as and increase ?
lout
lin original data
l'in original data, plus retransmitted data
finite shared output link buffers
69
Causes/costs of congestion scenario 3
lout
  • Another cost of congestion
  • when packet dropped, any upstream transmission
    capacity used for that packet was wasted!

70
Approaches towards congestion control
Two broad approaches towards congestion control
  • Network-assisted congestion control
  • routers provide feedback to end systems
  • single bit indicating congestion (SNA, DECbit,
    TCP/IP ECN, ATM)
  • explicit rate sender should send at
  • End-end congestion control
  • no explicit feedback from network
  • congestion inferred from end-system observed
    loss, delay
  • approach taken by TCP

71
Case study ATM ABR congestion control
  • ABR available bit rate
  • elastic service
  • if senders path underloaded
  • sender should use available bandwidth
  • if senders path congested
  • sender throttled to minimum guaranteed rate
  • RM (resource management) cells
  • sent by sender, interspersed with data cells
  • bits in RM cell set by switches
    (network-assisted)
  • NI bit no increase in rate (mild congestion)
  • CI bit congestion indication
  • RM cells returned to sender by receiver, with
    bits intact

72
Case study ATM ABR congestion control
  • two-byte ER (explicit rate) field in RM cell
  • congested switch may lower ER value in cell
  • sender send rate thus maximum supportable rate
    on path
  • EFCI bit in data cells set to 1 in congested
    switch
  • if data cell preceding RM cell has EFCI set,
    sender sets CI bit in returned RM cell

73
TCPs Approach
  • Sender regulates rate of transmission based on
    perceived network congestion
  • Must consider
  • How does TCP sender limit the rate?
  • How does TCP sender perceive congestion?
  • What algorithm should control rate based on
    perceived congestion?
  • Reno congestion control algorithm

74
TCPs Approach
  • How does TCP sender limit the rate?
  • Congestion window constrains sender rate
  • LastByteSent-LastByteAcked ? minCongWin,
    RcvWindow
  • Unacknowledged data from sender
  • congwin regulates congestion

75
TCPs Approach
  • How does TCP sender perceive congestion?
  • congestion
  • Timeout
  • 3 duplicate acks
  • Result smaller congestion window (slower rate)
  • no congestion
  • Received acks
  • Result larger congestion window (higher rate)

76
TCPs Approach
  • TCP Congestion Control Algorithm
  • Three mechanisms
  • Additive-increase, multiplicative-decrease
  • Slow start
  • Reaction to timeout events

77
Additive increase, multiplicative decrease (AIMD)
  • Approach increase transmission rate (window
    size), probing for usable bandwidth, until loss
    occurs
  • additive increase increase CongWin by 1 MSS
    every RTT until loss detected
  • multiplicative decrease cut CongWin in half
    after loss

Saw tooth behavior probing for bandwidth
congestion window size
time
Congestion Avoidance
78
TCP Slow Start
  • When connection begins, CongWin 1 MSS
  • Example MSS 500 bytes RTT 200 msec
  • initial rate 20 kbps
  • available bandwidth may be gtgt MSS/RTT
  • desirable to quickly ramp up to respectable rate
  • When connection begins, increase rate
    exponentially fast until first loss event
  • (double rate every RTT)

79
TCP Slow Start (more)
  • When connection begins, increase rate
    exponentially until first loss event
  • double CongWin every RTT
  • done by incrementing CongWin for every ACK
    received
  • Summary initial rate is slow but ramps up
    exponentially fast

Host A
Host B
one segment
RTT
two segments
four segments
80
Refinement
  • Q When should the exponential increase switch to
    linear?
  • A When CongWin gets to 1/2 of its value before
    timeout.
  • Implementation
  • Variable Threshold
  • At loss event, Threshold is set to 1/2 of CongWin
    just before loss event

81
Refinement inferring loss
  • After 3 dup ACKs
  • CongWin is cut in half
  • window then grows linearly
  • But after timeout event
  • CongWin instead set to 1 MSS
  • window then grows exponentially
  • to a threshold, then grows linearly

Philosophy
  • 3 dup ACKs indicates network capable of
    delivering some segments
  • timeout indicates a more alarming congestion
    scenario

82
Summary TCP Congestion Control
  • When CongWin is below Threshold, sender in
    slow-start phase, window grows exponentially.
  • When CongWin is above Threshold, sender is in
    congestion-avoidance phase, window grows
    linearly.
  • When a triple duplicate ACK occurs, Threshold set
    to CongWin/2 and CongWin set to Threshold.
  • When timeout occurs, Threshold set to CongWin/2
    and CongWin is set to 1 MSS.

83
TCP sender congestion control
State Event TCP Sender Action Commentary
Slow Start (SS) ACK receipt for previously unacked data CongWin CongWin MSS, If (CongWin gt Threshold) set state to Congestion Avoidance Resulting in a doubling of CongWin every RTT
Congestion Avoidance (CA) ACK receipt for previously unacked data CongWin CongWinMSS (MSS/CongWin) Additive increase, resulting in increase of CongWin by 1 MSS every RTT
SS or CA Loss event detected by triple duplicate ACK Threshold CongWin/2, CongWin Threshold, Set state to Congestion Avoidance Fast recovery, implementing multiplicative decrease. CongWin will not drop below 1 MSS.
SS or CA Timeout Threshold CongWin/2, CongWin 1 MSS, Set state to Slow Start Enter slow start
SS or CA Duplicate ACK Increment duplicate ACK count for segment being acked CongWin and Threshold not changed
84
TCP throughput
  • Whats the average throughout of TCP as a
    function of window size and RTT?
  • Ignore slow start
  • Let W be the window size when loss occurs.
  • When window is W, throughput is W/RTT
  • Just after loss, window drops to W/2, throughput
    to W/2RTT.
  • Average throughout .75 W/RTT

85
TCP Futures TCP over long, fat pipes
  • Example 1500 byte segments, 100ms RTT, want 10
    Gbps throughput
  • Requires window size W 83,333 in-flight
    segments
  • Throughput in terms of loss rate
  • ? L 2?10-10 Wow
  • New versions of TCP for high-speed needed!

86
TCP Fairness
  • Fairness goal if K TCP sessions share same
    bottleneck link of bandwidth R, each should have
    average rate of R/K

87
Why is TCP fair?
  • Two competing sessions
  • Additive increase gives slope of 1, as throughout
    increases
  • multiplicative decrease decreases throughput
    proportionally

R
equal bandwidth share
loss decrease window by factor of 2
congestion avoidance additive increase
Connection 2 throughput
loss decrease window by factor of 2
congestion avoidance additive increase
Connection 1 throughput
R
88
Fairness (more)
  • Fairness and parallel TCP connections
  • nothing prevents app from opening parallel
    connections between 2 hosts.
  • Web browsers do this
  • Example link of rate R supporting 9 cnctions
  • new app asks for 1 TCP, gets rate R/10
  • new app asks for 11 TCPs, gets R/2 !
  • Fairness and UDP
  • Multimedia apps often do not use TCP
  • do not want rate throttled by congestion control
  • Instead use UDP
  • pump audio/video at constant rate, tolerate
    packet loss
  • Research area TCP friendly

89
Delay modeling
  • Notation, assumptions
  • Assume one link between client and server of rate
    R
  • S MSS (bits)
  • O object size (bits)
  • no retransmissions (no loss, no corruption)
  • Window size
  • First assume fixed congestion window, W segments
  • Then dynamic window, modeling slow start
  • Q How long does it take to receive an object
    from a Web server after sending a request?
  • Ignoring congestion, delay is influenced by
  • TCP connection establishment
  • data transmission delay
  • slow start

90
Fixed congestion window (1)
  • First case
  • WS/R gt RTT S/R ACK for first segment in window
    returns before windows worth of data sent

delay 2RTT O/R
91
Fixed congestion window (2)
  • Second case
  • WS/R lt RTT S/R wait for ACK after sending
    windows worth of data sent

delay 2RTT O/R (K-1)S/R RTT - WS/R
92
TCP Delay Modeling Slow Start (1)
  • Now suppose window grows according to slow start
  • Will show that the delay for one object is

where P is the number of times TCP idles at
server
- where Q is the number of times the server
idles if the object were of infinite size. -
and K is the number of windows that cover the
object.
93
TCP Delay Modeling Slow Start (2)
  • Delay components
  • 2 RTT for connection estab and request
  • O/R to transmit object
  • time server idles due to slow start
  • Server idles P minK-1,Q times
  • Example
  • O/S 15 segments
  • K 4 windows
  • Q 2
  • P minK-1,Q 2
  • Server idles P2 times

94
TCP Delay Modeling (3)
95
TCP Delay Modeling (4)
Recall K number of windows that cover
object How do we calculate K ?
Calculation of Q, number of idles for
infinite-size object, is similar (see HW).
96
HTTP Modeling
  • Assume Web page consists of
  • 1 base HTML page (of size O bits)
  • M images (each of size O bits)
  • Non-persistent HTTP
  • M1 TCP connections in series
  • Response time (M1)O/R (M1)2RTT sum of
    idle times
  • Persistent HTTP
  • 2 RTT to request and receive base HTML file
  • 1 RTT to request and receive M images
  • Response time (M1)O/R 3RTT sum of idle
    times
  • Non-persistent HTTP with X parallel connections
  • Suppose M/X integer.
  • 1 TCP connection for base file
  • M/X sets of parallel connections for images.
  • Response time (M1)O/R (M/X 1)2RTT sum
    of idle times

97
HTTP Response time (in seconds)
RTT 100 msec, O 5 Kbytes, M10 and X5
For low bandwidth, connection response time
dominated by transmission time.
Persistent connections only give minor
improvement over parallel connections.
98
HTTP Response time (in seconds)
RTT 1 sec, O 5 Kbytes, M10 and X5
For larger RTT, response time dominated by TCP
establishment slow start delays. Persistent
connections now give important improvement
particularly in high delay?bandwidth networks.
99
Chapter 3 Summary
  • principles behind transport layer services
  • multiplexing, demultiplexing
  • reliable data transfer
  • flow control
  • congestion control
  • instantiation and implementation in the Internet
  • UDP
  • TCP
  • Next
  • leaving the network edge (application,
    transport layers)
  • into the network core
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